Protocol
Development
In the last ten years, the number of users on the Internet has increased at a dramatic rate with more and more applications
deployed in the data world and no end in sight. The intent of one of these applications was to carry voice over a data network; this eventually took the name
of Voice over Internet Protocol (VoIP). The initial VoIP deployments had two functions: to convert voice from/to data packets and es-tablish a protocol to
get the phones in contact with each other, similar to the switching concept in the TDM world.
As IP deployment continued its explosive
growth, VoIP became increasingly attractive, and the need for standards became obvious. With-out standards there was no hope of interoperability and user
features were limited to the basic Plain-Old-Telephone-Service (POTS). Also, administration of larger networks and security would become a
nightmare.
One of the first efforts at standardization resulted in the H.323 standard. H.323 is really an umbrella specification
encompassing many addi-tional specifications. The "H.xxx" signifies that this specification comes from ITU-T (formally CCITT). H.323 may be
efficient in terms of memory and CPU usage, but it is difficult for programmers to work with, and difficult to expand and develop. REDCOM, and the VoIP
industry in gen-eral, have embraced SIP as the best protocol for converged communication going forward based in ease of use, human readability and SIP's
ability to expand.
SIP Capabilities
Session
Initiation Protocol (SIP), in contrast to H.323, represents a complete break with the tradition of TDM telephony protocols as it comes from the data world.
SIP is not a specific protocol for telephony. Rather, SIP is a generalized protocol for allowing what are called "user agent" clients and
servers to associate, and when communication is desired, allow them to exchange capabilities, make media choices and establish communication sessions between
them. SIP can mediate a session involving a one-way transmission of a full-length movie, a multi-party video conference, a telephone call, or an Instant
Message transmission. SIP provides a set of methods and a suggested way to use them to create (in the case of VoIP) call flows for most of the commonly used
features we are familiar with in the TDM world. Just a few years ago, SIP was not rich enough to permit more than simple POTS telephone service without
giving up interoperability. For example, VoIP phones could not be mixed and matched.
There are almost daily improvements being made in the
SIP implementations as more and more vendors, including REDCOM, follow SIP standards. This increases interoperability between different vendor
equipment.
TRANSip Features
As
the convergence of TDM and IP networks continue, end users do not necessarily want to abandon their traditional telephony
features.
TRANSip's integrated architecture allows the migration of many rich traditional TDM features to standard SIP phones, but certain
specific fea-tures require support from the phone manufacturer (i.e. support for hook-flash). The following is a partial list of traditional TDM features
that are supported by TRANSip.
Selected Features Available to SIP phones
Line
Groups: SIP phones, along with analog and digital phones, can be put in a line group to benefit from group call pick-up, broadcast ringing,
speed dial, and many others.
Station Number Playback: SIP phone users dial a code and receive an announcement
that relays the number of the phone they are dialing from.
Caller ID and Name: Caller ID and name features
available to the analog and digital phones are ported to the SIP phones along with the ability to choose not sending Caller ID and name, as well as blocking
calls based on Caller ID.
Announcements: Common announcements can be sent to analog, digital and even SIP
phones. This ensures common announcements for all of the users.
Call Detail Records: Traditional call detail
records (CDR) that were logged for billing or quality assurance reasons are also available for SIP phones. TRANSip provides records including callED number,
callING number, call time, call duration and other detailed information for SIP phones as well as analog and ISDN phones.
Call hold, call
pickup, call waiting and call return features can all be accomplished in the SIP phones that support hook-flash functionality.
Compatible SIP Phones
In the traditional TDM world, the phone or the end station was easily compatible
with the existing phone lines. The backward compatibility is so strong that an operational phone from the early 20th Century can be connected to the existing
telephone network.
As the intelligence shifted to the end terminals in the IP world, the phones started to deploy new capabilities. However
there are still inter-operability issues as the phone vendors may deploy proprietary signaling protocols. In order to ease the concern of its customers,
REDCOM periodically tests various phones in the industry.
REDCOM has
tested a number of SIP phones for compatibility with TRANSip. However, the only SIP phones certified for use with DoD communicators are the
following:
- Cisco 7940 Series
- Cisco 7960 Series
